#include "simplest_ffmpeg_audio_player2.h"

/**
 * 最简单的基于FFmpeg的音频播放器  1.2
 * Simplest FFmpeg Audio Player  1.2
 *
 * 雷霄骅 Lei Xiaohua
 * leixiaohua1020@126.com
 * 中国传媒大学/数字电视技术
 * Communication University of China / Digital TV Technology
 * http://blog.csdn.net/leixiaohua1020
 *
 * 本程序实现了音频的解码和播放。
 *
 * This software decode and play audio streams.
 */

#include <stdlib.h>
#include <string.h>
#include "libavcodec/avcodec.h"
#include "libavformat/avformat.h"
#include "libswresample/swresample.h"
//SDL
#include "SDL/SDL.h"
#include "SDL/SDL_thread.h"

#define MAX_AUDIO_FRAME_SIZE 192000 // 1 second of 48khz 32bit audio


//Output PCM
#define OUTPUT_PCM 1
//Use SDL
#define USE_SDL 1

//Buffer:
//|-----------|-------------|
//chunk-------pos---len-----|
static  Uint8  *audio_chunk;
static  Uint32  audio_len;
static  Uint8  *audio_pos;

/* The audio function callback takes the following parameters:
 * stream: A pointer to the audio buffer to be filled
 * len: The length (in bytes) of the audio buffer
 * 回调函数
 */
void  fill_audio2(void *udata, Uint8 *stream, int len){
    if(audio_len==0)        /*  Only  play  if  we  have  data  left  */
        return;
    len=(len>audio_len?audio_len:len);    /*  Mix  as  much  data  as  possible  */
    
    SDL_MixAudio(stream,audio_pos,len,SDL_MIX_MAXVOLUME);
    audio_pos += len;
    audio_len -= len;
}
//-----------------


int audio_player2(const char* url)
{
    AVFormatContext    *pFormatCtx;
    int                i, audioStream;
    AVCodecContext    *pCodecCtx;
    AVCodec            *pCodec;
    
    //char url[]="WavinFlag.aac";
    //char url[]="72bian.mp3";
    //char url[]="72bian.wma";
    
    av_register_all();
    avformat_network_init();
    pFormatCtx = avformat_alloc_context();
    //Open
    if(avformat_open_input(&pFormatCtx, url, NULL, NULL)!=0){
        printf("Couldn't open input stream.\n");
        return -1;
    } else {
        printf("open input stream successful.\n");
    }
    // Retrieve stream information
    if(avformat_find_stream_info(pFormatCtx, NULL)<0){
        printf("Couldn't find stream information.\n");
        return -1;
    }
    // Dump valid information onto standard error
    av_dump_format(pFormatCtx, 0, url, 0);
    
    // Find the first audio stream
    audioStream=-1;
    for(i=0; i < pFormatCtx->nb_streams; i++)
        if(pFormatCtx->streams[i]->codec->codec_type==AVMEDIA_TYPE_AUDIO){
            audioStream=i;
            break;
        }
    
    if(audioStream==-1){
        printf("Didn't find a audio stream.\n");
        return -1;
    }
    
    // Find the decoder for the audio stream
    pCodec=avcodec_find_decoder(pFormatCtx->streams[audioStream]->codecpar->codec_id);
    if(pCodec==NULL){
        printf("Codec not found.\n");
        return -1;
    }
    
    // Get a pointer to the codec contextint audio_player2() for the audio stream
    pCodecCtx=pFormatCtx->streams[audioStream]->codec;
    
    // Open codec
    if(avcodec_open2(pCodecCtx, pCodec,NULL)<0){
        printf("Could not open codec.\n");
        return -1;
    } else {
        printf("open codec successful.\n");
    }
    
    FILE *pFile=NULL;
#if OUTPUT_PCM
    pFile=fopen("output.pcm", "wb");
#endif
    
    AVPacket *packet=(AVPacket *)malloc(sizeof(AVPacket));
    av_init_packet(packet);
    
    //Out Audio Param
    // 立体声双通道类型
    uint64_t out_channel_layout = AV_CH_LAYOUT_STEREO;
    int out_nb_samples = 1024;
    //参数三：out_sample_fmt->输出采样精度->编码
    enum AVSampleFormat out_sample_fmt = AV_SAMPLE_FMT_S16;
//    enum AVSampleFormat out_sample_fmt = pCodecCtx->sample_fmt;;
    //参数四：out_sample_rate->输出采样率(44100HZ)
    int out_sample_rate= pCodecCtx->sample_rate;
    int out_channels=av_get_channel_layout_nb_channels(out_channel_layout);
    //Out Buffer Size
    int out_buffer_size=av_samples_get_buffer_size(NULL,out_channels, out_nb_samples, out_sample_fmt, 1);
    
    uint8_t *out_buffer=(uint8_t *)av_malloc(MAX_AUDIO_FRAME_SIZE*2);
    
    AVFrame    *pFrame;
    pFrame=av_frame_alloc();
    //SDL------------------
#if USE_SDL
    //Init
    if(SDL_Init(SDL_INIT_VIDEO | SDL_INIT_AUDIO | SDL_INIT_TIMER)) {
        printf( "Could not initialize SDL - %s\n", SDL_GetError());
        return -1;
    }
    // SDL_AudioSpec
    SDL_AudioSpec wanted_spec;
    wanted_spec.freq = out_sample_rate;
    wanted_spec.format = AUDIO_S16SYS;
    wanted_spec.channels = out_channels;
    wanted_spec.silence = 0;
    wanted_spec.samples = out_nb_samples;
    wanted_spec.callback = fill_audio2;
    wanted_spec.userdata = pCodecCtx;
    
    if (SDL_OpenAudio(&wanted_spec, NULL)<0){
        printf("can't open audio.\n");
        return -1;
    }
#endif
    printf("Bitrate:\t %3d\n", (int)pFormatCtx->bit_rate);
    printf("Decoder Name:\t %s\n", pCodecCtx->codec->long_name);
    printf("Channels:\t %d\n", pCodecCtx->channels);
    printf("Sample per Second\t %d \n", pCodecCtx->sample_rate);
    
    int ret,len = 0;
    int got_picture;
    int index = 0;

    //Swr
    struct SwrContext *au_convert_ctx;
    au_convert_ctx = swr_alloc();
    int64_t in_channel_layout = av_get_default_channel_layout(pCodecCtx->channels);
    /*
     * @param s               需要设置参数的SwrContext对象
     * @param out_ch_layout   输出通道的布局 (AV_CH_LAYOUT_*)
     * @param out_sample_fmt  输出采样格式 (AV_SAMPLE_FMT_*).
     * @param out_sample_rate 输出采样率 (frequency in Hz)
     * @param in_ch_layout    输入通道的布局 (AV_CH_LAYOUT_*)
     * @param in_sample_fmt   输入采样格式 (AV_SAMPLE_FMT_*).
     * @param in_sample_rate  输入采样率 (frequency in Hz)
     * @param log_offset      日志
     * @param log_ctx         日志上下文对象，可以为NULL
     */
    au_convert_ctx=swr_alloc_set_opts(au_convert_ctx,
                                      out_channel_layout,
                                      out_sample_fmt,
                                      out_sample_rate,
                                      in_channel_layout,
                                      pCodecCtx->sample_fmt,
                                      pCodecCtx->sample_rate,
                                      0,
                                      NULL);
//    au_convert_ctx=swr_alloc_set_opts(au_convert_ctx,
//                                      out_channel_layout,
//                                      pCodecCtx->sample_fmt,
//                                      pCodecCtx->sample_rate,
//                                      av_get_default_channel_layout(pCodecCtx->channels),
//                                      pCodecCtx->sample_fmt,
//                                      pCodecCtx->sample_rate,
//                                      0,
//                                      NULL);
    swr_init(au_convert_ctx);
    while(av_read_frame(pFormatCtx, packet)>=0){
        if(packet->stream_index==audioStream){
            ret = avcodec_decode_audio4( pCodecCtx, pFrame, &got_picture, packet);
//            avcodec_send_packet(pCodecCtx, packet);
//            avcodec_receive_frame(pCodecCtx, pFrame);
            if ( ret < 0 ) {
                printf("Error in decoding audio frame.\n");
                return -1;
            }
            if ( got_picture > 0 ){
                swr_convert(au_convert_ctx,
                            &out_buffer,
                            MAX_AUDIO_FRAME_SIZE,
                            (const uint8_t **)pFrame->data,
                            pFrame->nb_samples);
                
                //FIX:FLAC,MP3,AAC Different number of samples
                if(wanted_spec.samples!=pFrame->nb_samples){
                    SDL_CloseAudio();
                    out_nb_samples=pFrame->nb_samples;
                    out_buffer_size=av_samples_get_buffer_size(NULL,out_channels ,out_nb_samples,out_sample_fmt, 1);
                    wanted_spec.samples=out_nb_samples;
                    SDL_OpenAudio(&wanted_spec, NULL);
                }
                
#if OUTPUT_PCM
                //Write PCM
                fwrite(out_buffer, 1, out_buffer_size, pFile);
#endif
                
                index++;
            }
            
            //SDL------------------
#if USE_SDL
            //Set audio buffer (PCM data)
            audio_chunk = (Uint8 *)out_buffer;
            //Audio buffer length
            audio_len =out_buffer_size;
            
            audio_pos = audio_chunk;
            //Play
            SDL_PauseAudio(0);
            while(audio_len>0)//Wait until finish
                SDL_Delay(1);
#endif
        }
        av_packet_unref(packet);
    }
    
    swr_free(&au_convert_ctx);
    
#if USE_SDL
    SDL_CloseAudio();//Close SDL
    SDL_Quit();
#endif
    // Close file
#if OUTPUT_PCM
    fclose(pFile);
#endif
    av_free(out_buffer);
    // Close the codec
    avcodec_close(pCodecCtx);
    // Close the video file
    avformat_close_input(&pFormatCtx);
    
    return 0;
}
